[asterisk-users] Caller ID on Transfers

Douglas Garstang dgarstang at oneeighty.com
Tue Jul 25 12:31:13 MST 2006


> -----Original Message-----
> From: Joshua Colp [mailto:jcolp at digium.com]
> Sent: Tuesday, July 25, 2006 8:54 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
> 
> 
> ----- Original Message -----
> From: Douglas Garstang
> [mailto:dgarstang at oneeighty.com]
> To: Asterisk Users Mailing List -
> Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com]
> Sent:
> Tue, 25 Jul 2006 15:37:10 -0300
> Subject: RE: [asterisk-users] Caller ID on
> Transfers
> 
> > > 
> > > What type of transfer? blind or attended?
> > 
> > Does it matter? Both...
> > 
> > Doug.
> 
> Yes, it does matter. An attended SIP transfer is handled much 
> differently then a blind transfer. It starts out as a regular 
> call to another person, Asterisk doesn't know it's actually a 
> transfer. Then when the transfer actually happens the phone 
> says "hey, this call I have up over here to you... it's 
> replacing this other call".

I thought the new SIP invite had a 'Diverted' field or something in it? If that's true, this is really bad because I need some way in my AGI script to determine that it's a transferred call, and not a new call. In the case of a transferred call, we want to set the caller id to the original calling party info, not the transferring party info. I swear that last week when I was doing this, the RDNIS agi variable was being set and I could use that to set the caller id information as needed. However, now it's suddenly stopped working and I don't know why.

> 
> Call flow:
> 
> Call #1
> Phone A ---> Asterisk ---> Phone B
> (Phone A performs attended transfer)
> Phone B is put on hold.
> 
> Call #2
> Phone A ---> Asterisk ---> Phone C
> (Phone A transfers Phone B to Phone C)
> 
> Transfer
> "Hey Asterisk, this call with ID 4653456sdfgawe45 I have up 
> to Phone B... it's replacing this call adsf8wet I have up to 
> Phone C -- they should be talking to eachother"
> Phone B ---> Asterisk ---> Phone C
> Phone A disappears out of both calls.
> 
> With a blind transfer the phone can simply say hey channel... 
> this is your new extension and context.

Is this documented somewhere?



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